1. Technical Field
The present invention relates generally to a communications system and in particular to a method and system for improving flow of data traffic within a communications network. Still more particularly, the present invention relates to a method and system for improving flow of data traffic within a multimedia communications network by reducing congestion.
2. Description of the Related Art
The H.323 standard is an umbrella recommendation from the International Telecommunication Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that do not provide a guaranteed Quality of Service (QoS). These networks dominate today""s corporate desktops and include packet-switched Transmission Control Protocol/Internet Protocol (TCP/IP) and Internet Packet Exchange (IPX) over Ethernet, Fast Ethernet and Token Ring network technologies. Therefore, the H.323 standard is an important building block for a broad new range of collaborative, LAN-based applications for multimedia communications.
The H.323 standard is the newest member of a family of ITU umbrella recommendations which cover video telephone multimedia communications over a variety of pipelines. The H.323 standard is in many senses a derivative of H.320, 1990 umbrella recommendation for video telephone over switched digital telephone networks. The H.323 standard borrows heavily from H.320""s structure, modularity, and audio/video compression/ decompression (codec) recommendations.
The H.323 standard provides a foundation for audio, video, and data communications across IP based networks, including the Internet. By complying to the H.323 standard, multimedia products and applications from multiple vendors can interoperate, allowing users to communicate without concern for compatibility. The H.323 standard will be the keystone for LAN based products for consumer, business, entertainment, and professional applications.
Communications under the H.323 standard can be considered a mix of audio, video, and control signals. Audio capabilities, Q.931 call setup, RAS control, and H.245 signaling are required. All other capabilities including video and data conferencing are optional. When multiple algorithms are possible, the algorithm utilized by the encoder are derived from information passed by the decoder during the H.245 capability exchange. H.323 terminals are also capable of asymmetric operation (different encode and decode algorithm) and can send/receive more than one video and audio channel.
The H.323 standard addresses call control, multimedia management, and bandwidth management for point-to-point and multipoint conferences. It is designed to run on common network architectures. As network technology evolves, and as bandwidth management techniques improve, H.323-based solutions will be able to take advantage of the enhanced capabilities. The H.323 standard is not tied to any hardware or operating system and H.323-compliant platforms will be available in all sizes and shapes, including video-enabled personal computers, dedicated platforms, and turnkey boxes.
Often, users want to conference without worrying about compatibility at the receiving point. The H.323 standard establishes standards for compression and decompression of audio and video data streams, ensuring that equipment from different vendors will have some area of common support. Besides ensuring the receiver can decompress the information, the H.323 standard establishes methods for receiving clients to communicate capabilities to the sender. The standard also establishes common call setup and control protocols.
The H.323 standard utilizes both reliable and unreliable communications. Control signals and data require reliable transport because the signal must be received in the order in which they were sent and cannot be listed. Audio and video streams lose their value with time. If a packet is delayed, it may not have relevance to the end user. Audio and video signals utilize the more efficient but less reliable transport.
Because the H.323 standard is Real-Time Transport Protocol (RTP) based, it can operate on the Internet""s Multicast Backbone (Mbone), a virtual network on top of the Internet that provides a multicast facility, and supports video, voice and data conferencing. The H.323 [H.323v2] standard has been proposed to perform call control (i.e. make connections) of real-time service on IP networks. The H.323 standard allows end-points or terminals wanting to make connections to negotiate bandwidth and coding requirements before the connection is established. In this standard there are three key players:
End-point: These are terminals which need to make connections. They request the connection through a gatekeeper (if one is on the network) and they also negotiate the connection parameters.
Gatekeeper: These entities perform bandwidth control (on LANs) and routing of connection packets towards the destination terminal.
Gateway: This entity can be thought of as a collection of end-points, but these entities also translate from other bearer protocols (such as time-division multiplexing (TDM)) to the IP protocol.
FIG. 1 clearly illustrates the interconnectivity of these components. FIG. 1 depicts a network with several gatekeepers 100, routers 101, endpoints 102, gateways 108, and terminals 104. Gatekeepers 100, routers 101, gateways 108, endpoints 102, and terminals 104 are interconnected via network links 106. Note that gatekeepers 100 are linked together to form the framework of the network while gateways 108, endpoints 102 and terminals 104 serve as the branches to this framework.
The Gatekeeper is a H.323 entity that provides address translation, control access, and sometimes bandwidth management to the LAN for H.323 terminals, Gateways, and Multipoint Control Units (MCUs). Gatekeepers perform two important call control functions which help preserve the integrity of the corporate data network. The first is address translation from LAN aliases for terminals and gateways to IPX addresses, as defined in the Registration/Admission/Status (RAS) specification. The second function is bandwidth management, which is also designated within RAS. For instance, if a network manager has specified a threshold for the number of simultaneous conferences on the LAN, the Gatekeeper can refuse to make any more connections once threshold is reached. The effect is to limit the total conferencing bandwidth to some fraction of the total available, the remaining capacity is left for email, file transfers, and other LAN protocols. The collection of all Terminals, Gateways and Multipoint Control Units managed by a single gatekeeper is known as a H.323 Zone.
Improvements in communications arise from changing user""s needs and demands. Previously, public network needs were driven by telephoning and voice data. Data traffic has grown slowly until recently. With the lower cost in telecommunications and the higher increase in processing power of computers, the number of users accessing communications networks has increased. The needs of these users include, for example, video telephone, low cost video conferencing, imaging, high definition television (HDTV), and other applications requiring multimedia data transfers. Multimedia combines different forms of media in the communication of information between a user and a data processing system, such as a personal computer. A multimedia application is an application that utilizes different forms of communications within a single application. Multimedia applications may, for example, communicate data to a user on a computer via audio, text, and video simultaneously. Such multimedia applications are usually bit intensive, real time, and very demanding on communications networks.
The H.323 standard sets multimedia standards for the existing infrastructure (i.e. IP-based networks). Design to compensate for the effect of highly variable LAN latency, the H.323 standard allows customers to utilize multimedia applications without changing their network infrastructure.
Reliable transmission of messages utilizes a connection-oriented mode for data transmission. Reliable transmission guarantees sequenced error-free, flow-controlled transmission of packets, but can delay transmission and reduce throughput. The H.323 standard utilizes reliable (TCP) end-to-end service for the H.245 Control Channel, the T.120 Data Channel and the Call Signaling Channel.
Within the IP stack, unreliable services are provided by User Datagram Protocol (UDP). Unreliable transmission is a mode without connection with promises nothing more than xe2x80x9cbest effortxe2x80x9d delivery. UDP offers minimal control information. It is a network layer which sits at the same level of network stack as TCP. It is a connection-less protocol within TCP/IP that corresponds to the transport layer in the ISO/OSI model. UDP converts data messages generated by an application into packets to be sent via IP but does not verify that messages have been delivered correctly. The H.323 standard utilizes UDP for the audio, video and the RAS Channel.
IP networks are the technology driving the Internet. The rise of these networks is primarily due to their acceptance as the layer 3 protocol in the enterprise networks. Most PCs now utilize transmission control protocol/Internet protocol (TCP/IP) as their networking protocol. IP has even gained acceptance as the wide area protocol since it is about 25-30% more efficient than ATM.
The kinds of traffic running over IP networks are of two major types:
Elastic traffic or non-real-traffic which is primarily data file transfer. Most of this traffic uses TCP as its transport level protocol and it can withstand delay quite well, but any corruption of data must be re-transmitted; and
The inelastic or real-time traffic is interactive voice, video or data-conferencing. This kind of traffic does not withstand delay well since late information in an interactive session is of no use. This kind of traffic utilizes real time protocol (RTP) over UDP as the transport protocol.
UDP is the dominant multimedia protocol. However, it does not have any inherent congestion control mechanism. A need thus exists for Real-Time Transport Control Protocol (RTCP) protocol on top of UDP to control delays. RTCP works with RTP for multimedia services.
Running real-time traffic over IP network has other significant problems also. Currently, there is no way of reserving bandwidth end-to-end in an IP network. Each IP packet takes its own route through the network. Therefore, each packet gets to its destination (in theory) through a different route and can have a different delay in getting to its destination. This causes delay variance or jitter at the destination where the packets have to be xe2x80x9cplayedxe2x80x9d for the destination user.
There have been some concerns in voice-over-IP (VoIP) industry that the introduction of large volume voice traffic into an IP network will unfairly compete for network bandwidth with existing TCP traffic. TCP has congestion control mechanisms built in. Once TCP senses network congestion by its detection of lost packets, it will reduce its packet transmission rate. Therefore, in case of network congestion, all TCP connections will throttle back until the congestion is relieved. However, UDP does not have similar control mechanisms. For now, UDP traffic in IP networks has been minimal. Although only TCP traffic reacts to network congestion, it has not been a problem. It is expected that the introduction of VoIP services will bring in a large volume of UPD traffic. Voice UDP traffic is an ill-behaved source and can potentially lock out TCP traffic in case of congestion. Since other multi-media services, such as video conferencing, are also expected to use UDP as the transport layer protocol, this problem exits for all IP multi-media services. Data applications currently use and will continue to use reliable transmission protocols (i.e. TCP) because data integrity is the top priority. The perceived UDP traffic increase will come from IP multimedia applications. Some congestion control mechanism is required to manage multi-media UDP traffic.
Therefore, it would be desirable to have an improved method for reducing congestion in the flow of data traffic in a multimedia communications network. Additionally, it would be desirable to reduce such congestion flow without significant interruption in the flow of data within the multimedia communications network.
It is one object of the present invention to provide an improved method and system for a communications system.
It is another object of the present invention to provide an improved method and system for improving flow of data traffic within a communications network.
It is yet another object of the present invention to provide an improved method and system for improving flow of data traffic within a multimedia communications network by reducing congestion.
The above features are achieved as follows. A method is disclosed for reducing congestion of real time data traffic on a multimedia communications network having a traffic control mechanism. The method comprises first extracting from data traffic in the multimedia communications network information regarding congestion of the multimedia communications network. Secondly, congestion is regulated on the multimedia communications network utilizing the network information extracted from the multimedia communications network.
In accordance with a preferred embodiment of the present invention, a plurality of monitors scans the through data traffic for RTCP packets. The RTCP packets provide information on the traffic flow which is extracted by the monitors. The information is forwarded to a central server where it is analyzed. Following this analysis, the central server initiates steps to relieve congestion in the network.
The above as well as additional objectives, features, and advantages of the present invention will become apparent in the following detailed written description.